Multimedia and group communications have become an important aspect of telecommunications, and the demand for such continues to increase. For instance, the Final Report of the Public Safety Wireless Advisory Committee to the Federal Communications Committee (“FCC”), dated 1996, expressed the critical need for communication resources for multimedia. Subsequently in 1998, the FCC established a band plan for the 764 MHz frequencies that included spectrum set aside for public safety wideband. In addition, the Internet Engineering Task Force (“IETF”) has developed a suite of protocols that are designed for use in multimedia communications. These protocols include a Session Initiation Protocol (“SIP”), a Session Announcement Protocol (“SAP”), and a Session Description Protocol (“SDP”).
Since its approval in early 1999 as an official standard, SIP has gained tremendous market acceptance for signaling communications services on the Internet. As such, numerous products incorporate the SIP standard, including but not limited to SIP desktop telephones, SIP telephony servers, and personal computing (“PC”) devices running SIP applications. SIP is a text-based signaling transactional protocol, similar to Hypertext Transfer Protocol (“HTTP”) and Simple Mail Transfer Protocol (“SMTP”), and works in the Application layer of the Open Systems Interconnection (“OSI”) communications model. A SIP message is used to control interactive communications sessions, such as voice, video, and chat, between users (also referred to herein as callers) in a communications network. Each user is typically associated with a communications device (also referred to herein as a terminal device) that is connected to the network.
SIP was designed for controlling media sessions and for establishing media sessions between an initiating endpoint and one recipient endpoint or a small group of recipient endpoints. However, SIP is not readily scalable for establishing media sessions between an initiating endpoint and a large group of recipient endpoints. This is because in standard SIP, three messages (INVITE/OK/ACK) must be exchanged between the initiating endpoint and each recipient endpoint in a given group. If a group is particularly large, this excessive messaging could cause bandwidth and timing problems, which is not desirable for communications that are time sensitive, e.g., as in the area of public safety.
Known systems for group communication have attempted to use standard SIP for enabling group communications. To do this, these systems have implemented a call control architecture in accordance with a server-centric architecture 100 illustrated in FIG. 1. Architecture 100 may be included in a push-to-talk (PTT) communications system. Architecture 100 includes a service specific server 102 that may be, for instance a PTT server, communicatively coupled in the signaling path of an endpoint 104 and a group 110 comprising endpoints 112, 114, and 116.
Using this paradigm, group communication is supported by PTT server 102, which is known to endpoints 104, 112, 114, and 116 as the target for all call control signaling. To setup a session, an initiating endpoint must target a session request to the PTT server 102 by using its Internet Protocol (IP) address. Specifically, while the call control signaling utilized by in the session request may identify the group in some manner, routing to the PTT server 102 is accomplished by performing a domain name system (DNS) lookup and using network layer IP protocols. This approach is limited in that it ties group communication to specific servers, thereby, limiting a system's ability to perform load balancing and failure recovery and placing an additional burden on an initiating endpoint by requiring it to know not only the group name but the IP address of the correct server.
In addition, existing group communications approaches have limited scalability and performance because a distinct SIP call leg must be used to join each group member to a session. Thus, as the number of group members increases, more and more three-way signaling exchanges must be performed over shared wire-line and wireless links before session setup can be completed. For large groups, the serialization delays can increase call setup times beyond what is acceptable for certain systems, especially public safety dispatch systems.
One example of a system that uses SIP signaling in the above described limiting fashion for group communications is the system disclosed in WO0167674 A2 (Ser. No. 09/518,776), entitled METHOD AND APPARATUS FOR PARTICIPATING IN GROUP COMMUNICATION SERVICES IN AN EXISTING COMMUNICATION SYSTEM. The specification describes a PTT server that can be added to a packet data system to provide Voice Over IP (VoIP) group communication. When a terminal desires to communicate with a net (a group of endpoints), the terminal determines the IP address of the appropriate top level server using DNS to resolve the SIP server addresses into Internet network addresses, and sends a SIP INVITE to this server requesting a session with the net. This server is further the target of call control signaling, talker arbitration signaling, and media. In addition, only point-to-point SIP signaling is used.
Thus, there exists a need for a call control architecture that takes advantage of the application layer routing control of SIP to enable a user to initiate a session based only the name and domain of the target regardless of which particular terminals they happen to be using, but that further includes features to enable greater scalability and faster call set-up for large groups of users.